Existing delta-sigma analog to digital converters (ADCs) contain an analog noise-shaping loop, followed by a digital decimation filter which suppresses the out-of-band quantization error and permits a reduced clock frequency close to the Nyquist rate. Often, the analog loop and the digital filter are integrated on the same chip. However, in some specific applications (e.g., in some digital audio systems) the analog loop and the decimation filter must be in different physical locations. In this case, the loop should operate with a 1-bit digital output to allow the use of a standard digital interface with high dynamic range. It should also have a nearly full-scale input range, which is hard to achieve with single-bit noise shaping.
A use of analog to digital (A/D) converters in digital audio apparatus is to carry out A/D conversion of the input sound and music signal from a microphone for digital media. However, output level of the microphone (which changes the air vibration from a sound source into an analog electricity signal) is extremely small. Even when amplified by an electric amplifier, the analog output signal level from a microphone is at most about few millivolts. The input full-scale level of an A/D converter with a signal to noise ratio (SNR) of 90 dB has a dynamic range of usually about several volts. In this case, a 10 dB to 40 dB amplifier is required between the microphone and ADC. Moreover, the distance between a microphone and sound source is not usually fixed. The above-mentioned amplifier normally has a variable gain amplifier which can change gain value according to the situation for optimal recording. Noise in the analog wiring from the microphone to the variable amplifier and the A/D converter is a significant problem.
Usually, a microphone is arranged near the sound source (such as man and a musical instrument) and a variable amplifier and A/D converter are arranged in a digital audio apparatus such as a recorder. The signal transmission between the microphone and digital apparatus is carried by analog wiring. Therefore, if a disturbance noise appears in this analog wiring, the noise will be amplified by the variable amplifier. This will be input to the A/D converter. This causes significant tone quality degradation.
U.S. Pat. Application Publication, Pub. No.: US 2007/0127761 (Pub. Date: Jun. 7, 2007), “Microphone Comprising Integral Multi-level Quantizer and Single bit Conversion Means”, discloses a digital microphone comprising an integral analog-to-digital converter based on a analog delta sigma modulator with multi-level quantizer in cascade with a digital signal converter which is adapted to provide a single-bit output signal. It focuses on a digital microphone application. It discloses two types of embodiments where the digital signal converter is adapted to provide a single-bit output signal. The first type is a single bit digital delta sigma converter of FIG. 2 in U.S. 2007/0127761, and the second type is direct symbol mapping method of FIG. 3 in U.S. 2007/0127761. But it does not describe analog performance and output signal duty ratio.
In the case of today's digital microphone application with single bit PDM signal output, several important target specifications are signal bandwidth, signal to noise ratio, oversampling ratio, and output signal duty ratio, etc. These are described later in Table 1. Oversampling Ratio is normally fixed to 64 times, that means over sampling rate is 64 times 48 kHz for the conventional DVD system, and 64 times 44.1 kHz for the Super Audio Compact Disc (SACD) system.
In the case of the direct symbol mapping method of FIG. 3 in U.S. 2007/0127761, the oversampling ratio of the multi-level analog delta sigma modulator should be expanded to double for tri-level, or quadruple for fifth-level, etc. This means that the oversampling ratio of the multi-level analog delta sigma modulator should be reduced to 32 times for tri-level quantizer, or to 16 times for fifth-level quantizer, to get the oversampling ratio of 64 times on the single bit output of the direct symbol mapping method. In the case of the delta sigma modulator, the signal to noise ratio will be drastically reduced by the reduction of oversampling ratio and cannot be compensated with such small levels as tri-level or fifth-level. Therefore, the direct symbol mapping method is not feasible to achieve a sufficient signal to noise ratio for digital microphone applications.
Another embodiment in U.S. 2007/0127761 is a third-order analog delta sigma modulator with multi-level quantizer of FIG. 4 in US 2007/0127761. It is cascaded with the same third-order single bit digital delta sigma converter of FIG. 2 in US 2007/0127761. In this embodiment, the loop order of both modulators is the same and the circuit topologies are designed as compatible with each other. The only difference is that the quantizer is multi-level versus single-bit. The signal to noise ratio of the total system will be limited by the single-bit digital delta sigma modulator and also by the chip area and power consumption for the third-order analog delta sigma modulator. These would be too much for the needed signal to noise ratio of total system. Additionally, there is no description or design consideration regarding the overload that will happen in the delta sigma modulator for large analog signal input. There is also no consideration regarding the duty ratio of the single-bit output signal. To get enough dynamic range on the single-bit output signal, the duty ratio would be near 90% or more for the maximum analog input level; this is more difficult in higher order delta sigma modulators. This means that the order and circuit topology should be separately considered so as to get best performance with low power consumption and cost for the digital microphone application.
U.S. Pat. No. 6,326,912 discloses an analog-to-digital converter comprising a front-end multi-bit delta sigma modulator coupled directly, or indirectly, to a back-end single-bit delta sigma modulator. The disclosed main application is a 1 bit stream format recordable system, Super Audio Compact Discs, used for DVD Audio systems or special format Audio CD recording systems. The disclosed embodiments are mainly focusing on the use of multi-bit analog delta sigma modulators including multi-stage cascaded topology for the application of 1 bit stream format of SACD. This uses a back-end 1 bit digital delta sigma modulator, and also focuses on a sampling rate conversion between both delta sigma modulators usable in the case of parallel usage with a conventional decimator to get a conventional Nyquist rate 16 to 24 bit PCM audio signal. Nyquist rate is 48 kHz in a DVD audio system and 44.1 kHz in a CD audio system. U.S. Pat. No. 6,326,912 includes comments about overload in the conventional 5th order single loop 1 bit delta-sigma modulator. It uses a gain scaling method to avoid overload and to get a stable higher-order loop. However, the gain scaling method will result in attenuation of the input signal, and then reduction of dynamic range of the 1 bit PDM output signal. That is, the duty ratio of the 1 bit PDM signal becomes 50% when the gain scaling is 50%. In U.S. Pat. No. 6,326,912, because this gain-scaled 1 bit delta sigma modulator with the duty ratio of the 1 bit PDM signal as low as 50% for the maximum analog input was already used in the main application of SACD, there was not enough consideration for the duty ratio of the 1 bit PDM signal. This is a very important specification for today's digital microphone application.
What is needed are techniques for separate analog loop and digital filter components that provide high dynamic, near full scale range performance for use with a standard digital interface. The above so-called analog microphone particularly has a need to mitigate noise in the analog signal line. The hybrid delta-sigma modulator described solves such problems.